Hi, we currently have the following setup.
A single trunk with 25 channels
5 locations with their own DDI (5 DDIs)
Is it possible to limit each DDI to only use 5 channels so each location will always have a 5 channels available?
Thanks
2 posts - 2 participants
Hi,
I have a CyberData 011410 SIP intercom that I recently gave its own extension on my FreePBX 16.0.40.7 setup. I was looking to have it call several extensions on the system when the button is pressed, so I made a ring group (ext 1003) with those extentions (ext 202, 203, etc) and had the device dial that. The call appears on all of the VoIP phones (a Poly Rove B2 in this case) , but answering on any of them results in no audio coming out of the intercom. I do hear audio on my B2 coming from the intercom’s microphone. If I make the intercom dial one of the extensions directly, then everything works as it should. All of these devices are on the same local network, so no VPN/NAT issues are here.
I assume I have something configured wrong, so any advice would be appreciated. Doing a search on the forums shows an old thread with a similar issue, but it was locked before any answers were posted. I have linked two pastebins logs of me dialing and answering the call. Nothing in particular sticks out to me between them that shows the issues, but maybe it will to an expert hanging around here 
If I need to provide any more information, I can do so.
Thanks!
1 post - 1 participant
I select iso ( version 2306-1) from freepbxdistro , put it in usb drive (rufus4.4), and then 
:
1 post - 1 participant
just as the title suggests i can only acces freepbx after running from console fwconsole firewall stop, the same thing happens with accessing the ssh, till yesterday i could access everything normally but now i cant, is there smth i have config wrong or should check?
Thank you in advance
4 posts - 2 participants
I have been using twillio with freepbx for a while but never noticed our CID was not working…
We emailed Twilio and got the CID added on there… But we still get null on the CID when we call out.
has anyone had the problem before and how do you fix it.
Not really sure what-else we need to do to get it working
8 posts - 2 participants
Hello everyone.
I have two freepbx installed in two different location, with two different sip account, interconnected using iax2. On the second pbx I have free calls flat rate and I would to use that possibility on the first too.
It’s possible to send all outgoing calls from first pbx to the outbound route of the second one trough iax2 connection? If yes, how can I get this?
Thanks for your help!
Joe
1 post - 1 participant
Hi
when using a softphone app on a mobile, contacts are often stored with +countrycode number.
When choosing a contact to call using the app, the +sign is forwarded to the PBX.
Is there a smart way to define the outbound route pattern and indicate “00” or “+” instead of duplicating any row.
thx
Marc
2 posts - 2 participants
I’ve been tasked with standing up a new Asterisk server running FreePBX v16.0.40.7. My existing Asterisk is on FreePBX v2.11.0.38. It seems obvious to me that I can’t simply run a backup on the old one and restore it on the new one given the massive version mismatch. I’m hesitant to try to update the existing server (if it’s even possible at this point) to get a backup from a similar version as the new server is on just because I don’t want to risk bricking the existing server before the new one is ready. Can anybody share advice on how I should proceed?
Apologies for the vagueness of my post. I’m about as green as it gets with Asterisk/FreePBX.
Thanks in advance.
Evan
1 post - 1 participant
Has anyone been successful in connecting FreePBX to Microsoft Teams and Voip.ms yet?
There’s a great tutorial by [Voxtel](https://voxtelesys.com/tutorial/free-pbx-teams-setup) which show us how to connect. There’s also Vodia. They created a great video. Has anyone been able to do this yet?
1 post - 1 participant
Running FreePBX 16 and having an annoying issue with inbound calls using my Voip.MS trunks. The calls will randomly not go through, but then be fine minutes later. The registration continues to show as active even when calls are failing, but nonetheless, calls fail.
Here’s a pastebin containing the asterisk logfiles for a recent example of an inbound call hitting the PBX but failing. NOTE: I have changed the DID to 0000000000 for privacy reasons - it was properly displaying the DID in the logs.
1 post - 1 participant
Hello
Firstly i would like to thanks @lgaetz @dicko @david55 @comtech @Stewart1 for all the help and support right from setting up the freepbx (from nov 22, till date) and ironing out multiple issues.
all is working fine since then…
now i have started using a call file bash script to automate outgoing calls and wombat (cheap way of getting call centre like facility).
so my few issues
as soon as the call from freepbx is transferred to matrix PBX, the recorded file starts playing even before the call has been actually answered. so one person can pick the call in 2 rings and another in 8 rings and thus the recording heard by the caller starts at different starting point.
in CDR of the freepbx i am getting “anonymous” as the caller ID. link to the thread
I guess both the issues are due to some misconfiguration in the matrix PBX.
My setup
FreePBX - 192.168.1.81
CO/PSTN Matrix Gateway -192.168.1.240 with 4 FXO and 4 FXS ports
“BSNL WINGS” SIP trunk service (India)
Pfsense + firewall with NAT and firewall rules
Dynamic public IP allocation by ISP - attached to DDNS domain by pfsense.
softphone on android and on iOS
MATRIX PBX SETTINGS
fxo port general settings for normal PSTN line
fxo port general settings for Fibre line
SIP TRUNK SETTINGS
SIP TRUNK SETTINGS
SIP TRUNK SETTINGS
SIP TRUNK SETTINGS
1 post - 1 participant
Hi,
I’ve set up FreePBX few days ago and i’m fighting with configuration.
Configured PJSIP and I see its working, when I call to PBX extension is calling, but on mobile phone i still see “calling” - even when I pick up call.
When we’re calling beetwen extensions everything is working.
When trying to call outbound I got info “The number you have dialed is not in service” - like extenstion dont have idea how to get to SIP Trunk.
Provider told Us to configure:
Provider:·
SIP: XXX, UDP port 5994
Media – RTP: XXXX, TCP/UDP port 30000-39999
FreePBX:
IP: XXX, UDP port 5061
Media – RTP: XXX, TCP/UDP port 8000-40000
Can somebody tell me what I’m doing wrong?
On incoming call I can see in asterisk info:
I can see RTP is sending/reciving when I did loggin.
On firewall I have enabled NAT on outgoing and VirutalIP to forward ports
1 post - 1 participant
Hi everyone,
we recently are experiencing an issue with fop2 address book in CID superfecta lookup. It seems that in the last version is not working anymore due to fop2 changes in database.
Anybody experienced the same issue and found a solution?
Thank you in advance for your help 
1 post - 1 participant
Hi all,
I set up our telephony with Issabel, switching from Elastix. Unfortunately, Issabel seems to be dead, so I would like to migrate to FreePBX in the not too distant future. I set up a virtual machine with v17 and it seems to work quite well so far and I really like to upgrades over v16 (and I’m a huge Debian-fan anyways). But now I’m wondering how to transfer my settings over to a FreePBX install. I found a couple of old threads here but they all link to a Wiki-page that no longer seems to exist. Does anyone have a tutorial that works?
TIA
SoWhy
1 post - 1 participant
Is there an overview of how the Directory Module is supposed to work?
I created a Misc App that goes to the directory but nothing is returning when I try and search
When Created the directory I set it as default, its the only one. All of the entries have greyed out text and are set to “voicemail greeting” I tried TTS and Spell name but nothing seems to be working.
Maybe I am going about it completely wrong
My end goal is a directory that someone can Dial into and search based on extension name. Speech recognition would also be slick but dial is fine for now.
4 posts - 2 participants
Hello. I am a total noob at this so forgive me if I am saying stuff that sounds stupid.
So, I’ve been following guides online to set up my own phone system. I have a raspberry pi connected to the service provider Flowroute which is connected to Freepbx and has an extension. I have Microsip as the SIP client.
In Microsip account settings according to the guides I’m supposed to input the settings like this:
Account Name: (any name)
SIP Server: (my raspberry pi ip address)
Username: (my extension number)
Domain: (my raspberry pi ip address)
Login: (my extension number)
password: (my freepbx extension password)
However, when I do this it times out and can’t seem to connect. The only way I’ve gotten my phone system to work at all is by putting the account settings in Microsip to connect directly to Flowroute. The problem with this is that when I call someone the CID is always unknown. I’m guessing this is because I’m bypassing the freepbx with these settings. I would like to set up my system so that I can change the CID from unknown to whatever number I want for my business. What do you think might be wrong with my setup?
2 posts - 2 participants
Hi,
Is there any way to assign automatically new SSL cert to webrtc extension in freepbx
2 posts - 2 participants
Hey ! Sorry to bother if this has a rather simple solution but I have been scratching my head for the past days over this error. I already deleted and reinstalled everything but it still doesnt work.
Quick explanation to my setup: I have a VPS where FreePBX is running on (Ubuntu). Requests to the IP Adress of the Server go through a Caddy Docker first and depending on the Domain are sent to apache (Listening on Port 50000) to then show FreePBX GUI.
Here is a screenshot of what the asterisk logs say when applying the configuration and trying to call the SIP Number:
The Inbound call problem isnt that much of a problem right now but I need to figure out where the problem within FreePBX/Asterisk is because if i set up a trunk and then check in SSH ithe Trunk is always “Unavailable”.
This is part of whats happening everytime i apply a configuration:
[2024-03-07 16:26:02] NOTICE[138525] app_queue.c: queuerules.conf has not changed since it was last loaded. Not taking any action.
[2024-03-07 16:26:20] ERROR[138020] res_pjsip.c: Error 320047 'No answer record in the DNS response (PJLIB_UTIL_EDNSNOANSWERREC)' sending OPTIONS request to endpoint Sebastian_trunk
[2024-03-07 16:26:59] WARNING[139152] db.c: AstDB key /TRUNK/1/dialopts does not exist
[2024-03-07 16:26:59] WARNING[139152] db.c: AstDB key /TRUNK/1/dialopts does not exist
[2024-03-07 16:27:02] Asterisk 20.6.0 built by root @ ubuntu on a x86_64 running Linux on 2024-03-07 12:53:33 UTC
[2024-03-07 16:27:02] WARNING[139382] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found and no default driver specified
[2024-03-07 16:27:02] NOTICE[139382] res_odbc.c: Registered ODBC class 'asteriskcdrdb' dsn->[MySQL-asteriskcdrdb]
[2024-03-07 16:27:02] NOTICE[139382] res_config_ldap.c: No directory user found, anonymous binding as default.
[2024-03-07 16:27:02] ERROR[139382] res_config_ldap.c: No directory URL or host found.
[2024-03-07 16:27:02] NOTICE[139382] res_config_ldap.c: Cannot reload LDAP RealTime driver.
[2024-03-07 16:27:02] WARNING[139382] res_config_pgsql.c: PostgreSQL RealTime: Not connected
[2024-03-07 16:27:02] NOTICE[139382] cdr.c: CDR logging disabled.
[2024-03-07 16:27:02] NOTICE[139382] indications.c: Default country for indication tones: us
[2024-03-07 16:27:02] NOTICE[139382] indications.c: Setting default indication country to 'us'
[2024-03-07 16:27:02] NOTICE[138020] res_pjsip/config_transport.c: Transport '0.0.0.0-udp' is not fully reloadable, not reloading: protocol, bind, TLS (everything but certificate and private key if filename is unchanged), TCP, ToS, or C>[2024-03-07 16:27:02] NOTICE[138020] sorcery.c: Type 'system' is not reloadable, maintaining previous values
[2024-03-07 16:27:02] WARNING[139382] res_phoneprov.c: Unable to find a valid server address or name.
[2024-03-07 16:27:02] ERROR[139382] ari/config.c: No configured users for ARI
[2024-03-07 16:27:02] NOTICE[139382] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[2024-03-07 16:27:02] NOTICE[139382] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[2024-03-07 16:27:02] WARNING[139382] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found and no default driver specified
[2024-03-07 16:27:02] WARNING[139382] cel_odbc.c: No such connection 'asteriskcdrdb' in the 'cel' section of cel_odbc.conf. Check res_odbc.conf.
[2024-03-07 16:27:02] WARNING[139382] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found and no default driver specified
[2024-03-07 16:27:02] WARNING[139382] cdr_adaptive_odbc.c: No such connection 'asteriskcdrdb' in the 'asteriskcdrdb' section of cdr_adaptive_odbc.conf. Check res_odbc.conf.
There are a lot of more errors and warnings but i guess it all traces back to some DNS problems caused by having multiple applications running on the same server.
Really REALLY glad if someone could help <3
1 post - 1 participant
Good afternoon!
We are setting up monday.com as our CRM and we are looking for ways to get the most of out it (and the tools that we currently are using).
We have FreePBX and we are using ClearlyIP desk phones.
We will be setting up information in monday.com that shows a telephone number. Is there a way that when one of our users clicks the telephone number that it can be sent to that user’s desk phone?
1 post - 1 participant